/* snd_alsa_0_9.c Support for ALSA 0.9 sound driver (cvs development version) Copyright (C) 1999,2000 contributors of the QuakeForge project Please see the file "AUTHORS" for a list of contributors This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to: Free Software Foundation, Inc. 59 Temple Place - Suite 330 Boston, MA 02111-1307, USA */ #ifdef HAVE_CONFIG_H # include "config.h" #endif static __attribute__ ((unused)) const char rcsid[] = "$Id$"; #include #include #include #include "QF/cvar.h" #include "QF/plugin.h" #include "QF/qargs.h" #include "QF/sound.h" #include "QF/sys.h" static int snd_inited; static snd_pcm_t *pcm; static const char *pcmname = NULL; static size_t buffer_size; static int snd_blocked = 0; static volatile dma_t sn; static cvar_t *snd_stereo; static cvar_t *snd_rate; static cvar_t *snd_device; static cvar_t *snd_bits; static plugin_t plugin_info; static plugin_data_t plugin_info_data; static plugin_funcs_t plugin_info_funcs; static general_data_t plugin_info_general_data; static general_funcs_t plugin_info_general_funcs; static snd_output_data_t plugin_info_snd_output_data; static snd_output_funcs_t plugin_info_snd_output_funcs; static void *alsa_handle; #define QF_ALSA_NEED(ret, func, params) \ static ret (*qf##func) params; #include "alsa_funcs_list.h" #undef QF_ALSA_NEED static qboolean load_libasound (void) { if (!(alsa_handle = dlopen ("libasound.so.2", RTLD_GLOBAL | RTLD_NOW))) { Sys_Printf ("couldn't load libasound.so.2: %s\n", dlerror ()); return false; } #define QF_ALSA_NEED(ret, func, params) \ if (!(qf##func = dlsym (alsa_handle, #func))) { \ Sys_Printf ("Couldn't load ALSA function %s\n", #func); \ dlclose (alsa_handle); \ alsa_handle = 0; \ return false; \ } #include "alsa_funcs_list.h" #undef QF_ALSA_NEED return true; } #define snd_pcm_hw_params_sizeof qfsnd_pcm_hw_params_sizeof #define snd_pcm_sw_params_sizeof qfsnd_pcm_sw_params_sizeof static void SNDDMA_Init_Cvars (void) { snd_stereo = Cvar_Get ("snd_stereo", "1", CVAR_ROM, NULL, "sound stereo output"); snd_rate = Cvar_Get ("snd_rate", "0", CVAR_ROM, NULL, "sound playback rate. 0 is system default"); snd_device = Cvar_Get ("snd_device", "", CVAR_ROM, NULL, "sound device. \"\" is system default"); snd_bits = Cvar_Get ("snd_bits", "0", CVAR_ROM, NULL, "sound sample depth. 0 is system default"); } static int SNDDMA_GetDMAPos (void); static qboolean SNDDMA_Init (void) { int err, frag_size; int rate = -1, bps = -1, stereo = -1; snd_pcm_hw_params_t *hw; snd_pcm_sw_params_t *sw; if (!load_libasound ()) return false; snd_pcm_hw_params_alloca (&hw); snd_pcm_sw_params_alloca (&sw); if (snd_device->string[0]) pcmname = snd_device->string; if (snd_bits->int_val) { bps = snd_bits->int_val; if (bps != 16 && bps != 8) { Sys_Printf ("Error: invalid sample bits: %d\n", bps); return 0; } } if (snd_rate->int_val) { rate = snd_rate->int_val; if (rate != 44100 && rate != 22050 && rate != 11025) { Sys_Printf ("Error: invalid sample rate: %d\n", rate); return 0; } } stereo = snd_stereo->int_val; if (!pcmname) pcmname = "default"; if ((err = qfsnd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) { Sys_Printf ("Error: audio open error: %s\n", qfsnd_strerror (err)); return 0; } Sys_Printf ("Using PCM %s.\n", pcmname); qfsnd_pcm_hw_params_any (pcm, hw); switch (rate) { case -1: if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw, 44100, 0) >= 0) { frag_size = 256; /* assuming stereo 8 bit */ rate = 44100; } else if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw, 22050, 0) >= 0) { frag_size = 128; /* assuming stereo 8 bit */ rate = 22050; } else if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw, 11025, 0) >= 0) { frag_size = 64; /* assuming stereo 8 bit */ rate = 11025; } else { Sys_Printf ("ALSA: no useable rates\n"); goto error; } break; case 11025: case 22050: case 44100: if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw, rate, 0) >= 0) { frag_size = 64 * rate / 11025; /* assuming stereo 8 bit */ break; } /* Fall through */ default: Sys_Printf ("ALSA: desired rate not supported\n"); goto error; } switch (bps) { case -1: if (qfsnd_pcm_hw_params_set_format (pcm, hw, SND_PCM_FORMAT_S16_LE) >= 0) { bps = 16; } else if (qfsnd_pcm_hw_params_set_format (pcm, hw, SND_PCM_FORMAT_U8) >= 0) { bps = 8; } else { Sys_Printf ("ALSA: no useable formats\n"); goto error; } break; case 8: case 16: if (qfsnd_pcm_hw_params_set_format (pcm, hw, bps == 8 ? SND_PCM_FORMAT_U8 : SND_PCM_FORMAT_S16) >= 0) { break; } /* Fall through */ default: Sys_Printf ("ALSA: desired format not supported\n"); goto error; } if (qfsnd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) { Sys_Printf ("ALSA: interleaved is not supported\n"); goto error; } switch (stereo) { case -1: if (qfsnd_pcm_hw_params_set_channels (pcm, hw, 2) >= 0) { stereo = 1; } else if (qfsnd_pcm_hw_params_set_channels (pcm, hw, 1) >= 0) { stereo = 0; } else { Sys_Printf ("ALSA: no useable channels\n"); goto error; } break; case 0: case 1: if (qfsnd_pcm_hw_params_set_channels (pcm, hw, stereo ? 2 : 1) >= 0) break; /* Fall through */ default: Sys_Printf ("ALSA: desired channels not supported\n"); goto error; } qfsnd_pcm_hw_params_set_period_size_near (pcm, hw, frag_size, 0); err = qfsnd_pcm_hw_params (pcm, hw); if (err < 0) { Sys_Printf ("ALSA: unable to install hw params\n"); goto error; } qfsnd_pcm_sw_params_current (pcm, sw); qfsnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U); qfsnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U); err = qfsnd_pcm_sw_params (pcm, sw); if (err < 0) { Sys_Printf ("ALSA: unable to install sw params\n"); goto error; } shm = &sn; memset ((dma_t *) shm, 0, sizeof (*shm)); shm->splitbuffer = 0; shm->channels = stereo + 1; shm->submission_chunk = qfsnd_pcm_hw_params_get_period_size (hw, 0); // don't // mix less than this # shm->samplepos = 0; // in mono samples shm->samplebits = bps; buffer_size = qfsnd_pcm_hw_params_get_buffer_size (hw); shm->samples = buffer_size * shm->channels; // mono samples in buffer shm->speed = rate; SNDDMA_GetDMAPos ();//XXX sets shm->buffer Sys_Printf ("%5d stereo\n", shm->channels - 1); Sys_Printf ("%5d samples\n", shm->samples); Sys_Printf ("%5d samplepos\n", shm->samplepos); Sys_Printf ("%5d samplebits\n", shm->samplebits); Sys_Printf ("%5d submission_chunk\n", shm->submission_chunk); Sys_Printf ("%5d speed\n", shm->speed); Sys_Printf ("0x%x dma buffer\n", (int) shm->buffer); snd_inited = 1; return 1; error: qfsnd_pcm_close (pcm); return 0; } static int SNDDMA_GetDMAPos (void) { const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t offset; snd_pcm_uframes_t nframes = shm->samples/shm->channels; if (!snd_inited) return 0; qfsnd_pcm_avail_update (pcm); qfsnd_pcm_mmap_begin (pcm, &areas, &offset, &nframes); offset *= shm->channels; nframes *= shm->channels; shm->samplepos = offset; shm->buffer = areas->addr;//XXX FIXME there's an area per channel return shm->samplepos; } static void SNDDMA_Shutdown (void) { if (snd_inited) { qfsnd_pcm_close (pcm); snd_inited = 0; } } /* SNDDMA_Submit Send sound to device if buffer isn't really the dma buffer */ static void SNDDMA_Submit (void) { int state; int count = (*plugin_info_snd_output_data.paintedtime - *plugin_info_snd_output_data.soundtime); const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t nframes; snd_pcm_uframes_t offset; if (snd_blocked) return; nframes = count / shm->channels; qfsnd_pcm_avail_update (pcm); qfsnd_pcm_mmap_begin (pcm, &areas, &offset, &nframes); state = qfsnd_pcm_state (pcm); switch (state) { case SND_PCM_STATE_PREPARED: qfsnd_pcm_mmap_commit (pcm, offset, nframes); qfsnd_pcm_start (pcm); break; case SND_PCM_STATE_RUNNING: qfsnd_pcm_mmap_commit (pcm, offset, nframes); break; default: break; } } static void SNDDMA_BlockSound (void) { if (snd_inited && ++snd_blocked == 1) qfsnd_pcm_pause (pcm, 1); } static void SNDDMA_UnblockSound (void) { if (!snd_inited || !snd_blocked) return; if (!--snd_blocked) qfsnd_pcm_pause (pcm, 0); } PLUGIN_INFO(snd_output, alsa0_9) { plugin_info.type = qfp_snd_output; plugin_info.api_version = QFPLUGIN_VERSION; plugin_info.plugin_version = "0.1"; plugin_info.description = "ALSA 0.9.x digital output"; plugin_info.copyright = "Copyright (C) 1996-1997 id Software, Inc.\n" "Copyright (C) 1999,2000,2001 contributors of the QuakeForge " "project\n" "Please see the file \"AUTHORS\" for a list of contributors"; plugin_info.functions = &plugin_info_funcs; plugin_info.data = &plugin_info_data; plugin_info_data.general = &plugin_info_general_data; plugin_info_data.input = NULL; plugin_info_data.snd_output = &plugin_info_snd_output_data; plugin_info_funcs.general = &plugin_info_general_funcs; plugin_info_funcs.input = NULL; plugin_info_funcs.snd_output = &plugin_info_snd_output_funcs; plugin_info_general_funcs.p_Init = SNDDMA_Init_Cvars; plugin_info_general_funcs.p_Shutdown = NULL; plugin_info_snd_output_funcs.pS_O_Init = SNDDMA_Init; plugin_info_snd_output_funcs.pS_O_Shutdown = SNDDMA_Shutdown; plugin_info_snd_output_funcs.pS_O_GetDMAPos = SNDDMA_GetDMAPos; plugin_info_snd_output_funcs.pS_O_Submit = SNDDMA_Submit; plugin_info_snd_output_funcs.pS_O_BlockSound = SNDDMA_BlockSound; plugin_info_snd_output_funcs.pS_O_UnblockSound = SNDDMA_UnblockSound; return &plugin_info; }